Sound Systems: Mono versus Stereo

It's a common question in many church sound system projects, "Will our system be mono or stereo?" What usually follows is a lengthy discussion about the applicability of mono sound systems versus stereo systems, the difference between two channel sound systems and stereo sound systems, the benefits and limitations of each, the architectural constraints and program requirements that will affect the decision, and the cost implications of each.

It is very apparent that everyone has their own interpretation of the terms 'mono' and 'stereo', influenced by their own experiences and expectations. Translating one's experience with home audio systems or project studios into a large venue like a church or a theatre often takes an adjustment in conceptual thinking, so we always have to provide a reference point for the discussion of mono, stereo and two channel sound systems. Let's start off with mono.

MouseOver - Single channel versus two channel loudspeaker coverage



Compare the single channel centre
cluster coverage to the following:

1. Single centre cluster
2. Both left & right speakers
3. Left channel speakers only 
4. Left channel with left "stinger"
5. Left channel with right "stinger"
6. Both channels with "stingers"

Mono

Mono or monophonic describes a system where all the audio signals are mixed together and routed through a single audio channel. Mono systems can have multiple loudspeakers, and even multiple widely separated loudspeakers. The key is that the signal contains no level and arrival time/phase information that would replicate or simulate directional cues. Common types of mono systems include single channel centre clusters, mono split cluster systems, and distributed loudspeaker systems with and without architectural delays. Mono systems can still be full-bandwidth and full-fidelity and are able to reinforce both voice and music effectively. The big advantage to mono is that everyone hears the very same signal, and, in properly designed systems, all listeners would hear the system at essentially the same sound level. This makes well-designed mono systems very well suited for speech reinforcement as they can provide excellent speech intelligibility.

Stereo

True stereophonic sound systems have two independent audio signal channels, and the signals that are reproduced have a specific level and phase relationship to each other so that when played back through a suitable reproduction system, there will be an apparent image of the original sound source. Stereo would be a requirement if there is a need to replicate the aural perspective and localization of instruments on a stage or platform, a very common requirement in performing arts centres.

This also means that a mono signal that is panned somewhere between the two channels does not have the requisite phase information to be a true stereophonic signal, although there can be a level difference between the two channels that simulates a position difference, this is a simulation only. That's a discussion that could warrant a couple of web pages all by itself.

An additional requirement of the stereo playback system is that the entire listening area must have equal coverage of both the left and right channels, at essentially equal levels. This is why your home stereo system has a "sweet spot" between the two loudspeakers, where the level differences and arrival time differences are small enough that the stereo image and localization are both maintained. This sweet spot is limited to a fairly small area between the two loudspeakers and when a listener is outside that area, the image collapses and only one or the other channel is heard. Living with this sweet spot in your living room may be OK, since you can put your couch there, but in a larger venue, like a church sanctuary or theatre auditorium, that sweet spot might only include 1/3 the audience, leaving 2/3 of the audience wondering why they only hear half the program.

In addition a stereo playback system must have the correct absolute phase response input to output for both channels. This means that a signal with a positive pressure waveform at the input to the system must have the same positive pressure waveform at the output of the system. So a drum, for instance, when struck produces a positive pressure waveform at the microphone and should produce a positive pressure waveform in the listening room. If you don't believe that this makes a tremendous difference, try reversing the polarity of both your hifi loudspeakers some day and listening to a source that has a strong centre sound image like a solo voice. When the absolute polarity is flipped the wrong way, you won't find a stable centre channel image, it will wander around away from the centre, localizing out at both the loudspeakers.

Two Channel

This is what many people mistake for stereo sound systems, because there are two channels and a "stereo" console is connected to the front of the system, and stereo amplifiers and equalizers are used throughout the system. What is missing from most of these systems is uniform coverage of the entire listening area, and a minimal level and phase response difference for each channel's coverage of the listening area. To achieve proper loudspeaker coverage to replicate a stereo image in a large venue, it is necessary to have a loudspeaker system for each channel that can provide uniform coverage of the entire listening area while maintaining the directional cues. This is a very expensive, and sometimes impossible proposition.

Two channel systems usually suffer from having half the people in the listening area only hear half the audio program, which makes two channel systems a poor choice for music reinforcement. A large portion of the listeners hear a completely different music mix from other listeners. This is an all-too-common oversight in venues that are intended for music and entertainment, even high profile venues where they deserve better system designs. It tends to be a common misconception brought forward by people with a background in portable or live sound systems.

When a two channel system is used to reinforce a mono voice microphone, the seats either side of the room centreline, exactly between the two loudspeakers, also experience substantial variations in frequency response and uniformity of coverage due to interference and signal cancellation when identical signals arrive from the two channels at different times. This makes two channel systems especially ineffectual for speech reinforcement applications.

Left/Centre/Right

There are specialized applications for sound systems described as Left/Centre/Right configurations. This must combine the best of both worlds, right? Well it can, but the loudspeaker system must also be designed to the highest common denominator not the lowest. It is also important with an LCR system that the mix engineer understand which signals must be routed to which loudspeakers, and which signal routings will create problems. LCR systems also are not suitable for use in all room shapes and configurations.

LCR systems are common in drama theatres and large churches where there is a requirement for both mono speech reinforcement and music or sound effects cues to be localized or mixed with a particular perspective, with a stereo or stereo-like imaging. The three loudspeaker systems must each provide coverage of the entire seating area while maintaining level and directional cues, just like the mono and stereo systems described above. There are some clever "cheats" a system designer can use to achieve the extended listening area for stereo coverage, and that invloves using "stingers" or infill loudspeakers.

Stingers

As shown above in the MouseOver demo, a stinger is a speaker positioned to provide coverage of a floor area that may be inaccessible from the left or right cluster mounting position.

In the example above, the left and right stingers are separate from the centre cluster, and would be fed from a signal delay so that for people in the right side of the room, the signal from the left stinger would arrive at the same time as the sound from the left cluster would have if it could have reached that location. For people sitting near the front right, they would still hear the program from the left channel to maintain the integrity of the program content, even though the stereo image would be skewed to one side compared to the image perceived by the people sitting in the middle of the room. The speaker selection, as well as the level and delay setting of the stingers are all quite critical to their successful integration, if they are too loud, too late or too early they will detract from the image the rest of the audience hears. If you're going to try this at home study up on Haas Effect, setting levels and delays for precedence, as well as time domain measurement systems.

It is also possible to use some of the components of the centre cluster as the stingers, especially with the advent of lower cost DSP systems that allow a matrix of signal delays to be developed for each input to the unit. Even a few years ago this would have been exceedingly expensive and complicated (and nearly impossible to explain), now it can be programmed quickly on a computer screen and be made to work quickly and easily. This approach works best when the centre cluster has similar sounding loudspeaker components to the left/right clusters.

Which one's the best?

As with many questions about sound systems, there is no one right answer. A well designed mono system will satisfy more people than a poorly designed or implemented two channel sound system. The important thing to keep in mind is that the best loudspeaker design for any facility is the one that will work effectively within the, programmatic, architectural and acoustical constraints of the room, and that means (to paraphrase the Rolling Stones) "You can't always get the system that you want, but you find some times that you get the system that you need." If the facility design (or budget) won't support an effective stereo playback or reinforcement system, then it is important that the sound system be designed to be as effective as possible, even if that means giving up a desirable program requirement like stereo.

Digital and Analog Mixers

Digital and Analog Mixers


The mixing console or "mixer" is a central component of most sound systems. In fact, the mixer used will have a large influence on the operability and efficiency of the entire system. One of the first decisions that will have to be made when designing a new sound system is whether a digital or analog mixer will be used.
Digital Mixers are Rapidly Gaining Ground
Described in the simplest terms, the difference between analog and digital mixers is whether audio signals are internally processed in their original analog form or converted to and processed in digital form. Digital mixers are rapidly gaining popularity for their convenience, expandability, and resistance to noise (see "The Merits of Digital Sound" in Part 1). But there are still situations in which analog is preferred. Let's take a look at the features of each.


Digital Mixer Features

Five of the most important features of digital mixers are listed below. All of these features are related to the benefits described inThe Merits of Digital Sound" in Part 1. It might be easier to grasp the "big picture" if you go back and review that information as well.
Settings can be pre-programmed and recalled when necessary
Most digital mixers feature some kind of memory into which settings can be stored and then instantly recalled whenever they are needed. This can be a tremendous advantage in a banquet hall facility for example, where the required settings might change frequently for different types of events and different room configurations. Even if some fine tuning is required, the ability to simply recall a complete set of basic parameters that are close to what is needed can dramatically reduce the time and effort required to set up for an event. And if a mistake is made, it's easy to revert to the basic settings.


 Mixing and processing features that only digital can provide

Digital technology has enabled the development of a number of mixing functions that were simply not available in analog systems. Automatic feedback suppression is one example. Automatic "ducking" that decreases the BGM level while an announcement is being made is another. More details will be provided in the "Automated Mix Functions" section.
Many digital mixers also feature built in signal processing functions such as effects that are designed to suit the mixer's intended applications. This makes it unnecessary to purchase extra external signal processing devices, thus significantly reducing overall system cost as well as installation space. This is a radical departure from analog systems for which additional equipment had to be purchased and installed to provide the signal processing functions required by each individual application. There are analog mixers that include some built-in processing functions, but unlike digital mixers in which all of the necessary processing can be implemented through software, additional processing capability has a direct influence on the cost and physical size of an analog mixer. For the same mixing and processing capabilities, a digital mixer will almost always offer superior economy.

Expansion and external devices

Expandable digital mixers make it possible to connect to a wide variety of external devices. Touch panel controllers, for example, can make day-to-day operation of a specified set of frequently used parameters easy for even inexperienced users, while keeping parameters that should not be changed hidden. More details will be provided in the "Peripheral Equipment" section.

Noise-resistant digital transmission

A sound system that is based on a digital mixer that uses digital transmission for audio signals will be highly resistant to induced noise, as described in the "The Merits of Digital Sound" section in Part 1. Since noise becomes more of a problem with longer transmission distances, the benefits of digital transmission in an audio installation increase as the size of the facility increases.

Multiple functions in small spaces

Although analog mixers usually have one control per function, in a digital mixer it is possible to assign numerous functions to a single control, with function switching either via physical controls or virtual controls on a display. This makes it possible to pack a large number of functions into a relatively small space. If you compare analog and digital mixers that offer the same number of channels, you'll see that the digital mixers tend to be significantly smaller.


Analog Mixer Features

Analog mixers have two main advantages.

* Lower cost for a limited set of features

If only a few channels with a basic set of mixing features are required, a simple analog mixer may be a more economical choice than a digital mixer.

*Easy operation for first-time users

As mentioned above, analog mixers usually have one control per function, all of which are visible and directly accessible via the control panel. The control layout logically follows the mixer's signal flow and is therefore relatively easy to understand. This type of logical, easy operation can be an advantage in public halls and schools, for example, where a variety of people, some having little or no previous experience, may need to operate the system. In such situations it may be necessary to protect controls and functions that should not be changed with security covers.
That covers the basic differences between digital and analog mixers. Depending on the application, the choice of a digital or analog mixer can make a large difference in the operability and overall economy of the system. Next we'll look at another important consideration for mixer selection: configuration.

Whats the Difference Between a DJ Mixer and a Live Mixer?

For a disc jockey or an audio technician, combining signals or sounds is both an art and a craft. These people know how to effectively direct and mix sounds so the audience can hear a harmonious output. Knowing how to wield a mixer, the tool of the trade, is the expertise of a professional disc jockey or audio technician. But even aspiring DJs or regular music lovers can learn how to use a mixer to create great music mixes.
There is a variety of mixers out on the market, and it is important to know what type of mixer is suitable for individual needs. A studio mixer, for instance, is a great tool for mixing sounds and recording a final output. Other types of mixers are a DJ mixer and a live mixer. Is there difference between the two mixers? Buyers must know the difference, as well as the different parts and functions of a mixer, in order to make a wise judgement when choosing the right type of mixer to use for either work or leisure.

Difference Between a DJ Mixer and a Live Mixer


All mixers have the same basic function: to combine signals. Mixers only vary because of their intended applications. Essentially, a mixer can either be a "live mixer" when it used in a live setting or a "studio mixer" when used for recording mixes and performances in a studio. A live mixer combines signals from different inputs and then sends it to a finite number of speakers, such as the main speakers, balcony speakers, and monitors. A studio mixer, on the other hand, combines signals to create a recorded master output.
Furthermore, a DJ mixer, which is mostly used in a live function, is in essence a live mixer. Usually, the two mixers both have the basic features, such as stereo channels, an input selector, a gain control, equalisers, and a microphone input. But what sets apart a DJ mixer from a typical live mixer is its ability to route a non-playing source to headphones, as well as the crossfader function that allows easier transition from one source to another.
Basic Parts of DJ and Live Mixer
Both live and DJ mixers have these basic parts: stereo channels, input selector, gain control, equaliser, cue switch, and microphone input. Although they are basic parts, they are important in the process of directing and mixing music during a live function, such as a party or a concert. A live or DJ mixer has two to six stereo channels for connecting source devices, such as a CD player, and sending the signals for mixing.
A DJ mixer has a phono input with RIAA equalisation, which is an essential specification if a DJ wants to attach a turntable to the mixer. Each channel is arranged vertically in both live and DJ mixers, and each channel has a knob or switch for selecting inputs. An equaliser is also present in any type of mixer so a DJ can fade in or fade out the different parts of a track. In mixing two tracks, one common technique is to mute the bass of one track and use only the other. This way, the bass lines of the two tracks do not clash.
There are mixers that feature separate knobs for the low, mid, and high frequency ranges, and controls like a balance knob, aux-sends for external effects unit, and built-in sound effects. A cue switch is important in a DJ mixer because it sends a signal to headphones, allowing a DJ to listen to that signal without it affecting the master output. At the end of a channel strip is a fader that is useful in varying the volume in the final mix. There are also microphone inputs in both a DJ mixer and a live mixer. Larger live mixing consoles have inputs for microphones and source devices to support a band setup.

Analogue Mixers vs. Digital Mixers


In the quest of finding the right DJ or live mixer, another question can come to mind: Should the mixer be analogue or digital? To have a better judgement in this matter, it is necessary to discuss the common desirable qualities in a mixer, such as fast signal processing, ease of use and great sound quality, as well as how an analogue or a digital mixer rate in these qualities.
Ease of Use
In terms of knob-, fader-, or button-function ratio, an analogue mixer wins over a digital mixing console. An analogue mixer has one knob, fader, or button per function, which makes it is easier and faster for a user to perform tasks on the console. A digital mixer, on the other hand, has fewer knobs, buttons, and faders; it has a simpler and clean look that does not intimidate a user.
Most digital mixers make up for the lack of physical controls with virtual pages or layers that change the fader banks into separate controls. The controls are useful in varying equalisation; however, this feature can be quite confusing for a beginning user. It is also harder to make a hardware patch change on a digital mixing console because of internal reassignment, making convenient groupings of inputs appear near each other at the fader bank. Modern digital mixers allow users to store and edit live mixes in a computer, which makes it easier for a DJ to create a remix.
Sound Quality
Both analogue and digital mixers make use of an analogue preamplifier that increases the strength of a signal at line level, so when it comes to sound quality, neither mixer beats the other. For overloading or clipping concerns, both mixers have a solution, with the digital mixer using a digital converter to break down an audio stream into small but measurable increments. The faint hissing sound to the main outputs can be corrected through good gain stage management in an analogue mixer and by low-level gating in a digital mixer.
Signal Processing
It is no secret that a digital mixer delays the release of a signal from the speakers because of the conversion of analogue signals to digital ones. The delay, depending on a digital mixer model and functions that are currently engaged in the process, ranges from 1.5 to 10 milliseconds. Although the delay is not a problem to someone hearing the output from loudspeakers, it is quite confusing and disorienting to an artist wearing in-ear monitors.
Remote Control Capability
Although analogue mixers has had the option of using a wired remote control since the 90s, it is the digital mixer that offers flexibility when it comes to remote controlling. Digital mixing platforms allow the use of both wired and wireless remote controls using a laptop or a tablet computer, especially if there is no need for quick changes in the mixing process during a live performance. There are computer networks linking different elements of a digital system to make it possible to manipulate devices even when they are far from the disc jockey or technician.
Buying a DJ or a Live Mixer on eBay
eBay is a great website to buy DJ mixers and live mixers because you have the option to choose from new, refurbished, and used mixers depending on your budget. Also, there are so many brand options for a DJ mixer or a live mixer on eBay, and you can easily filter your search to a certain brand that you fancy.
Searching on eBay is fast and easy. All you need to do is type the keywords "DJ mixers" or "live mixers" in the search bar to produce a listing of relevant results. You can also do a search based on features, such as an equaliser, USB connection, integrated effects unit, loop function, and more. Always remember to read the product description carefully to make sure that the specifications of a certain mixer is what you are really looking for. Also, check the shipping details of a product before purchasing to know when to expect it to arrive.

Conclusion

A quality mixer is any aspiring disc jockey’s dream. A live mixer or a DJ mixer, as opposed to a studio mixer, is usually a fixture in live events. It is such a versatile device; it allows not only mixing of tracks to produce a desirable output but also controlling of sounds that come out of the speakers to make an event more organized. Both a live mixer and a DJ mixer have the same basic parts, including the stereo channels, input selector, gain control, equaliser, cue switch, microphone input, and headphone input. Knowledge of the similarities and differences of a DJ mixer and a live mixer is essential in understanding their functions.
Moreover, it is also helpful to know the advantages and disadvantages of a digital DJ or live mixer and an analogue DJ or live mixer in order to know the best option for a particular purpose. Knowing a mixer inside and out may take years of experience, but knowing the different types of mixers and the basic parts are a first step to understanding how to maximise the functions of the device.

What Are the Differences Between All Those Audio Formats?

Digital audio has been around a very long time so there’s bound to be a plethora of audio formats out there. Here are some of the more common ones, what differentiates them, and what to use them for.

Before we talk about everyday audio formats, it’s important you understand the basics, and that means understanding PCM. After that, we’ll tackle compressed formats.

PCM Audio: Where It All Starts

Pulse-Code Modulation was created back in 1937 and is the closest approximation of analog audio. That is, an analog waveform is approximated in regular intervals. PCM is characterized by two properties: sample rate and bit depth. Sample rate measures how often (in times per second) the amplitude of the waveform is taken, and the bit depth measures the possible digital values. In terms of audio formats, this is pretty much the foundation.

True sound, in the real world, is continuous. In the digital world, it’s not. Somehow this is more confusing with audio than with video, so let’s look at video as a point of comparison. What we interpret to be “motion” or think of as “fluid” and constantly-moving is, in actuality, a series of still pictures. In that same way, the amplitude of sound waves in a digital format isn’t “fluid” or constantly changing. It’s changing based on certain criteria at pre-defined intervals.

I know there’s a lot here that may not be second-nature unless you’re an engineer, physicist, or an audiophile, so let’s pare it down further with an analogy.

Let’s say that the water flowing from an open faucet is your “analog” audio source. The temperature of the water we can compare to the amplitude of an audio wave; it’s a property that needs to be measured so you can enjoy it properly. Sampling is the number of times per second you dip your finger into the flowing water. The more often you dip your finger into it, the more “continuous” the temperature changes become. If you stick your finger into the running water 44,100 times per second, it’s almost like keeping your finger under there the whole time, right? That’s the basic idea behind sampling.

Bit depth is a little trickier. Instead of using your finger, let’s say you used a really crapper thermometer. It basically said “Hot” for anything above room temperature and “Cold” for anything below. Regardless of how many times you dipped it into the water, it wouldn’t really give you much useful information. Now, if instead of just 2 options, let’s say the thermometer had 16 possible values which you could use to gauge the water temperature. More useful, right? Bit depth works the same way, in that higher values allow more dynamic changes in sound amplitude to be accurately portrayed.

As previously mentioned, PCM is the foundation for digital audio, along with its variants. PCM attempts to model a waveform, in as much of its uncompressed glory as possible. It’s special, it’s ready to be stuck in a digital signal processor, and it’s more or less universally playable. Most other formats manipulate audio via algorithms, so they need to be decoded while playing. PCM audio is considered “lossless,” it is uncompressed, and therefore, takes up a lot of hard drive space.

The Uncompressed Bunch: WAV, AIFF

Both WAV and AIFF are lossless audio container formats based on PCM, with some minor changes in data storage. PCM audio, for most people, comes in these formats, depending on whether you use Windows or OS X, and they can be converted to and from each other without degradation of quality. They are both also considered “lossless,” are uncompressed, and a stereo (2-channel) PCM audio file, sampled at 44.1 kHz (or 44100 times per second) at 16 bits (“CD quality”) amounts to roughly 10 MB per minute. If you’re recording at home for the purposes of mixing, this is what you want to use because it’s full quality.

Lossless Formats: FLAC, ALAC, APE

The Free Lossless Audio Codec, Apple Lossless Audio Codec, and Monkey’s Audio are all formats which compress audio, much in the same fashion that anything is compressed in digital world: using algorithms. The difference between zipped files and FLAC files is that FLAC is designed specifically for audio, and so has better compression rates without any loss of data. Typically, you’re seeing about half the size of WAVs. That is, a FLAC file for stereo audio at “CD quality” runs roughly 5 MB per minute.

The up-side is that if you want to do audio manipulation, you can convert back to a WAV without any loss of quality. If you’re an audiophile and listen to a lot of music with dynamic ranges, these formats are for you. If you’ve got a great set of speakers, cans, or earbuds, these formats will bring out the tones to showcase them.

Lossy Formats: MP3, AAC, WMA, Vorbis

Most of the formats you see in day-to-day use are “lossy”; some degree of audio quality is sacrificed in exchange for a significant gain in file size. An average “CD quality” MP3 runs about 1 MB per minute. Big difference compared to PCM, no? This is called compression, but unlike with lossless formats, you can’t really get that quality back once you strip it in lossy formats. Different lossy formats use different algorithms to store data, and so they typically vary in file size for comparable quality. Lossy formats also use bitrate to refer to audio quality, which usually looks like “192 kbit/s” or “192 kbps.” Higher numbers means that more data is being pumped out, so there’s more preservation of detail. Here are some details for the more popular formats.

• MP3: MPEG 1 Audio Layer 3, the most common lossy audio codec today. Despite a heap of patent issues, it’s still incredibly popular. Who doesn’t have MP3s lying around?
• Vorbis: A free and open-source lossy format used more often in PC games such as Unreal Tournament 3. FOSS fans, such as many Linux users, are bound to see plenty of this format.
• AAC: Advanced Audio Coding, a standardized format now used with MPEG4 video. It’s heavily supported because of its compatibility with DRM (e.g. Apple’s FairPlay), its improvements over mp3, and because no license is needed in order to stream or distribute content in this format. Apple fans will probably have plenty in AAC.
• WMA: Windows Media Audio, Microsoft’s lossy audio format. It was developed and used to avoid licensing issues with the MP3 format, but because of major improvements and DRM compatibility, as well as a lossless implementation, it’s still around. It was really popular before iTunes became champion of DRMed music.

Lossy formats are what you use for all of the stuff you listen to and store. They’re designed to be an economy of hard drive space. Which format you choose depends on what digital audio player you use, how much space you have, how big of a quality nitpicker you are, and a bunch of over variables. Nowadays, computers will play anything, most audio players (except Apple’s, of course) will do multiple lossy formats, and more and more do FLAC and APE. Apple sticks to MP3, ALAC, and AAC.

Isn’t Audio Quality Subjective?

Absolutely, it is. Ultimately, it’s your ears that are consuming most of this stuff, but that’s more reason to think of quality seriously. When I first started creating my digital music collection, I couldn’t really tell the difference between 128kbit MP3s and audio CDs. To my ears, there was no noticeable difference. Over time, however, I noticed that 256 kbit sounded much better, and after I got a really nice (and expensive!) set of headphones, I went back to audio CDs full time! It also depends on the genre of music.

There are a LOT of variables here, folks, make no mistake about that. It took a while before I settled on using FLAC for some music and 320kbps MP3 for the rest. The point I’m trying to make is that you should experiment to see what works best for you and your music, but be aware that as your tastes change, your perceptions, your equipment, and the importance of quality will, too.

And all of this stuff get even trickier when you’re not just talking about music, but about voice tracks, sound effects, white and brown noise, etc. There’s a whole world of sound out there, so don’t get discouraged! By learning what you can and listening for yourself, you can use this info to your advantage in your future audio projects. I’ll leave you with some of the best advice I’ve ever gotten: “do what just plain sounds good.”

For every (EQ) action there is an equal and opposite (EQ) reaction

For example, if you add too much to the 2 kHz EQ band, eventually anything will sound thin and harsh. If you compensate by adding some 100 Hz to warm it up, you’ll end up with “scooped mids” and the sound will be thin and lack body. So you add some 500 and suddenly you’re back where you started, but it all sounds a bit processed and un-natural.

So I’ll finish with a final rule of thumb for you:

Less is more !

- and an outstanding resource, to an interactive frequency chart with even more rules of thumb and suggestions for the best EQ band to use eachinstrument. I don’t agree with all of them, but as Joe said in his video, there are no rules in audio – use your ears !

Instrument Frequency ranges and some crucial EQ bands and what they sound like

50-60 Hz

-Thump in a kick drum
-Boom in a bassline
-Essential in dub, dubstep and reggae !
-Too much and you’ll have flapping speakers and a flabby mix
-Too little, and the mix will never have enough weight or depth

100-200 Hz

-This EQ band adds punch in a snare
-Gives richness or “bloom” to almost anything
-Too much makes things boomy or woolly
-Too little sounds thin and cold

200-500 Hz

-Crucial for warmth and weight in guitars, piano and vocals
-Too much makes things sound muddy or congested
-Too little makes them thin and weak

500-1000 Hz

-One of the trickiest areas
-Gives body and tone to many instruments
-Too much sounds hollow, nasal or honky
-Too little sounds thin and harsh

2 kHz

-Gives edge and bite to guitars and vocals
-Adds aggression and clarity
-Too much is painful!
-Too little will sound soft or muted

5-10 kHz

-Adds clarity, open-ness and life
-Important for the top end of drums, especially snare
-Too much sounds gritty or scratchy
-Too little will lack presence and energy

16 kHz

-Can add air, space or sparkle
-Almost too high to hear
-Too much will sound artificial, hyped or fizzy
-Too little will sound dull and stifled

What is Audio mastering ?

Mastering, a form of audio post-production, is the process of preparing and transferring recorded audio from a source containing the final mix to a data storage device (the master); the source from which all copies will be produced (via methods such as pressing, duplication or replication). Recently, the format choice includes using digital masters although analog masters, such as audio tapes, are still being used by the manufacturing industry and by a few engineers who have chosen to specialize in analog mastering.

In order to make a deterministic process, mastering requires critical listening; therefore, it cannot be achieved without the presence of a mastering engineer. There are software mastering tools available to facilitate this last step, but results still depend upon the accuracy of speaker monitors. In addition, "music mastering" engineers may also need to apply corrective equalization and dynamic enhancement in order to improve upon sound translation on all playback systems.

>The studio<

The music mastering studio is very different from a normal audio recording studio. In fact, all the equipment and gear found in most recording and mixing studios can actually hinder the acoustics of a room to accurately monitor sound. Thus, the correct room acoustics and arrangement of the equipment inside a mastering studio is an important factor since the mastering engineer (ME) needs to hear each mix in detail. This room design should be non-environmental or with a minimum room interference. By working with an experienced mastering engineer, the recording artist is also open to alternative opinions and technical advice.

>Process<

The source material, ideally at the original resolution, is processed using equalization, compression, limiting, noise reduction and other processes. More tasks, such as editing, pre-gapping, leveling, fading in and out, noise reduction and other signal restoration and enhancement processes can be applied as part of the mastering stage. This step prepares the music for either digital or analog, e.g. vinyl, replication. The source material is put in the proper order, commonly referred to as assembly (or 'track') sequencing.

If the material is destined for vinyl release, additional processing, such as dynamic range reduction, frequency dependent stereo–to–mono fold-down and equalization, may be applied to compensate for the limitations of that medium. Finally, for compact disc release, Start of Track, End of Track, and Indexes are defined for disc navigation. Subsequently, it is rendered either to a physical medium, such as a CD-R or DVD-R, or to a DDP file set, the standard method of secure delivery for CD and DVD replication masters. The specific medium varies, depending on the intended release format of the final product. For digital audio releases, there is more than one possible master medium, chosen based on replication factory requirements or record label security concerns. Regardless of what delivery method is chosen, the replicator will transfer the audio to a glass master that will generate metal stampers for replication.

The process of audio mastering varies depending on the specific needs of the audio to be processed. Mastering engineers need to examine the types of input media, the expectations of the source producer or recipient, the limitations of the end medium and process the subject accordingly. General rules of thumb can rarely be applied.

Steps of the process typically include but are not limited to the following:

Transferring the recorded audio tracks into the Digital Audio Workstation (DAW) (optional).
Sequence the separate songs or tracks (the spaces in between) as they will appear on the final release.
Process or "sweeten" audio to maximize the sound quality for its particular medium (e.g. applying specific EQ for vinyl)
Transfer the audio to the final master format (i.e., CD-ROM, half-inch reel tape, PCM 1630 U-matic tape, etc.).

Examples of possible actions taken during mastering:

Editing minor flaws
Applying noise reduction to eliminate clicks, dropouts, hum and hiss
Adjusting stereo width
Adding ambience
Equalize audio across tracks
Adjust volume
Dynamic range expansion or compression
Peak limit